Index: webrtc/BUILD.gn |
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn |
index 3fb9cfea72ebf94abda5ea30f50f0f079ca32b94..65db50da9605896d159cb885b51b24f13afcf099 100644 |
--- a/webrtc/BUILD.gn |
+++ b/webrtc/BUILD.gn |
@@ -622,38 +622,15 @@ |
testonly = true |
configs += [ ":rtc_unittests_config" ] |
- sources = [ |
- "call/call_perf_tests.cc", |
- "call/rampup_tests.cc", |
- "call/rampup_tests.h", |
- "modules/audio_coding/codecs/opus/opus_complexity_unittest.cc", |
- "modules/audio_coding/neteq/test/neteq_performance_unittest.cc", |
- "modules/audio_processing/audio_processing_performance_unittest.cc", |
- "modules/audio_processing/level_controller/level_controller_complexity_unittest.cc", |
- "modules/audio_processing/residual_echo_detector_complexity_unittest.cc", |
- "modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc", |
- "video/full_stack_tests.cc", |
- ] |
deps = [ |
- "modules/audio_coding:neteq_test_support", |
- "modules/audio_processing", |
- "modules/audio_processing:audioproc_test_utils", |
- "modules/remote_bitrate_estimator:bwe_simulator_lib", |
- "modules/rtp_rtcp", |
- "test:test_common", |
+ "call:call_perf_tests", |
+ "modules/audio_coding:audio_coding_perf_tests", |
+ "modules/audio_processing:audio_processing_perf_tests", |
+ "modules/remote_bitrate_estimator:remote_bitrate_estimator_perf_tests", |
"test:test_main", |
- "test:test_renderer", |
+ "video:video_full_stack_tests", |
"video:video_quality_test", |
- "voice_engine", |
- "//testing/gmock", |
- "//testing/gtest", |
- ] |
- |
- if (rtc_enable_intelligibility_enhancer) { |
- defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ] |
- } else { |
- defines = [ "WEBRTC_INTELLIGIBILITY_ENHANCER=0" ] |
- } |
+ ] |
data = webrtc_perf_tests_resources |
if (is_android) { |
@@ -662,13 +639,6 @@ |
} |
if (is_ios) { |
deps += [ ":webrtc_perf_tests_bundle_data" ] |
- } |
- if (!build_with_chromium && is_clang) { |
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
- } |
- if (rtc_use_h264) { |
- defines += [ "WEBRTC_USE_H264" ] |
} |
} |